• 제목/요약/키워드: adaptive filter algorithm

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하다마드 변환을 이용한 적응필터의 특성 (Properties of Adaptive Filter Using Hadamard Transformation)

  • 이태훈;박진배
    • 제어로봇시스템학회:학술대회논문집
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    • 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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    • pp.242-242
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    • 2000
  • Comparing to the conventional adaptive filters using LMS algorithm, the proposed adaptive filters can reduce the amounts of computation and have robustness to variance of characteristics of input signals. LMS algorithm is performed in the domain of Hadamard transform after a reference signal and input signal are transformed by fast Hadamard transformation. As a transformation from time domain to Hadamard transformed domain, the proposed filter not only maintains the performance of estimating an input signal but also greatly reduces the number of multiplication. Moreover, the effect of characteristic changes of input signal is decreased. Computer simulation shows the stability and robustness of the proposed filter.

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DSP보드를 이용한 뇌파의 외부잡음 제거용 적응필터 및 피드백 출력제어 알고리듬 (The Adaptive Filter for EEG Artifact Cancellation and the Feedback Output Control Algorithm on the DSP Board)

  • 안보섭;박정제;이경일;박일용;조진호;김명남
    • 대한전기학회:학술대회논문집
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    • 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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    • pp.548-551
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    • 2003
  • The adaptive filter is proposed for removing EOG from measured EEG on the frontal lobe. The proposed adaptive filter has been implemented and the feedback output control algorithm has been employed to control the alpha wave ratio on the basis of TMS320C31 DSP board with the on-line and real time performance. The feedback algorithm controls the input voltage of stimulating devices on the portable bio-feedback system. The EEG data are acquired at the $F_{p1}$ and $F_{p2}$ localization and are processed by the proposed adaptive filter. We demonstrated that the proposed adaptive filter could effectively remove EOG from the measured EEG on the frontal lobe and the feedback algorithm is proper to control the output voltage of DSP board using the ratio of the alpha wave.

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ST세그먼트 검출성능향상을 종속 적응필터의 세계 (Design of a Cascade Adaptive Filter for the Performance sn Detection of Segment)

  • 박광리;이경중
    • 대한의용생체공학회:의공학회지
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    • 제16권4호
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    • pp.517-524
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    • 1995
  • This paper is a study on the design of the cascade adaptive filter (CAF) for baseline wandering elimination in order to enhance the performance of the detection of ST segments in ECG. The CAF using Least Mean Square (LMS) algorithm consists of two filters. The primary adaptive filter which has the cutoff frequency of 0.3Hz eliminates the baseline wandering in raw ECG The secondary adaptive filter removes the remnant baseline wandering which is not eliminated by the primary adaptive filter. The performance of the CAF was compared with the standard filter, the recursive filter, and the adaptive impulse correlated filter (AICF). As a result, the CAF showed a lower signal distortion than the standard filter and the AICF. Also, the CAF showed a better perf'ormance in noise elimination than the standard filter and the recursive filter. In conclusion, considering the characteristics of the noise elimination and the signal distortion, the CAF shows a better performance in the removal of the baseline wandering than the other three Otters and suggests the high performance in the detection of ST segment.

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슬라이딩 모드 관측기와 적응 필터를 이용한 SPMSM 기계 파라미터 추정 (SPMSM Mechanical Parameter Estimation Using Sliding-Mode Observer and Adaptive Filter)

  • 김형우;최준영
    • 전력전자학회논문지
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    • 제24권1호
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    • pp.33-39
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    • 2019
  • We propose a mechanical parameter estimation algorithm for surface-mounted permanent magnet synchronous motors (SPMSMs) using a sliding-mode observer (SMO) and an adaptive filter. The SMO estimates system disturbances in real time, which contain the information on mechanical parameters. A desirable feature that distinguishes the proposed estimation algorithm from other existing mechanical parameter estimators is that the adaptive filter estimates electromagnetic torque to improve the estimation performance. Moreover, the SMO acts as a low-pass filter to suppress the chattering effect, which enables the smooth output signals of the SMO. We verify the mechanical parameter estimation performance for SPMSM by conducting extensive experiments for the proposed algorithm.

Implementation of Multi-adaptive Filter for EOG Removal and Biofeedback Output Controller

  • Ahn, Bo-Sep;Kim, Pil-Un;Cho, Jin-Ho;Kim, Myoung-Nam
    • 한국멀티미디어학회논문지
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    • 제7권12호
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    • pp.1650-1656
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    • 2004
  • In this paper, a multi-adaptive filter is proposed for removing EOG and the 60 Hz power supply noise from EEG measured in the frontal lobe and the feedback output control method is implemented for biofeedback. The multi-adaptive filter has been implemented on the TMS320C6711 DSP system and the feedback output control algorithm has been realized by calculating the ratio of alpha wave on the TMS320C31 DSP system with real time performance. Through the experiment using the implemented multi-adaptive filter and feedback output controller, we demonstrate that the proposed adaptive filter effectively removes EOG and the 60 Hz power supply noise from the measured EEG in the frontal lobe and the feedback algorithm controls the level of stimulation by the ratio of the alpha wave.

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적응 L-필터의 수렴성 해석 (Convergence Analysis of Adaptive L-Filter)

  • 김수용;배성호
    • 한국멀티미디어학회논문지
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    • 제12권9호
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    • pp.1210-1216
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    • 2009
  • 본 논문에서는 순환최소순위(RLR) L-필터의 수렴성을 해석하였다. RLR L-필터는 순서통계필터로서 입력의 크기순서에 따른 가중치를 필터계수로 한다. 또한 RLR L-필터는 비선형 적응 필터로서 필터계수의 갱신을 위하여 RLR 알고리즘을 이용한다. RLR 알고리즘은 로버스트 통계학의 순위추정에 기초한 비선형 적응 알고리즘이다. 본 논문에서는 가변적인 스텝 크기를 적용하여 평균 및 평균제곱의 견지에서 수렴성을 해석하였다. RLRL-필터는 잡음의 분포함수가 두꺼운 꼬리 분포인 임펄스 잡음에 가까울수록 메디안 필터의 형태로 적응하며 가우시안 잡음의 경우 평균 필터의 형태로 적응한다.

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A convergence analysis of Block MADF algorithm for adaptive noise reduction

  • Min, Seung-gi;Young Huh;Yoon, Dal-hwan
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2002년도 ITC-CSCC -1
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    • pp.377-380
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    • 2002
  • When it calculates the optimum price of filter coefficient, the many operation quantity is necessary. Is like that the real-time control is difficult and the hardware embodiment expense is big. The case which does not know advance information of input signal or the case where the statistical nature changes with change of surroundings environment is necessary the adaptive filter. Every hour to change a coefficient automatically and system in order to reach to the condition of optimum oneself, the fact that is the adaptive filter. When it does not the quality of input signal or it does not know the environment of surroundings every hour changing, it does not emit not to be, in order to collect, the fact that is the adaptive filter. The case of the Acoustic Echo Canceler does thousands filter coefficients in necessity. It reduces a many calculation quantity to respect, it uses the IIR filter from hour territory. Also it uses the block adaptive filter which has a block input signal and a block output signal. The former there is a weak point where the stability discrimination is always demanded. Consequently, The block adaptive filter is researched plentifully. This dissertation planned the block MADF adaptive filter used to MADf algorithm.

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LMS ALGORITHM을 이용한 HYBRID CODING (HYBRID CODING USING THE LMS ALGORITHM)

  • 김승윈;이근영
    • 대한전기학회:학술대회논문집
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    • 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
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    • pp.1379-1382
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    • 1987
  • IN ADAPTIVE LINEAR PREDICTION, AN ADAPTIVE CAPABILITY IS BUILT INTO THE PROCESSOR SUCH THAT AS THE IMAGE STATISTICS CHANGE, THE PREDICTION FILTER COEFFICIENTS THEMSELVES CHANGE, PRODUCING A NEW FILTER MORE CLOSELY OPTIMIZED TO THE NEW SET OF IMAGES STATISTICS. THE LMS ALGORITHM MAY BE USED TO ADAPT THE COEFFICIENT OF AN ADAPTIVE PREDICTION FILTER FOR IMAGE SOURCE ENCODING. IN THIS PAPER, TWO CODING SYSTEMS USING DPCM AND LMS ALGORITHMS RESPECTIVELY FOR OBTAINING THE FIRST TRANSFORMED COEFFICIENT IN HYBRID CODING ARE COMPARED.

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적응 반향 제거기의 수렴 속도 향상 (Adaptive Echo Canceller with Improved Convergence Speed)

  • 김남선;임용훈;임종민;차일환;윤대희
    • 한국통신학회:학술대회논문집
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    • 한국통신학회 1991년도 추계종합학술발표회논문집
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    • pp.111-114
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    • 1991
  • This paper proposes an efficient adaptive echo canceller using pilot filter approach to achieve improved convergence speed. The pilot filter is an adaptive filter with only a few filter coefficients to filter the received signal for the purpose of whitening the signal. Thus the convergence speed of the main LMS-TDL filter combined with the pilot filter is improved. In the proposed echo canceller, an adaptive lattice predictor as the pilot filter is used and its inverse filter is used to equalize the distorted near end talker signal. Simulation results for colored signal show that the convergence speed of the proposed echo cancellation algorithm is faster than that of the conventional LMS-TDL echo cancellation algorithm.

추정상관값을 이용한 가변 스텝사이즈 LMS 알고리듬에 관한 연구 (A Study on Variable Step Size LMS Algorithm using estimated correlation)

  • 권순용;오신범;이채욱
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2000년도 추계종합학술대회 논문집(4)
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    • pp.115-118
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    • 2000
  • We present a new variable step size LMS algorithm using the correlation between reference input and error signal of adaptive filter. The proposed algorithm updates each weight of filter by different step size at same sample time. We applied this algorithm to adaptive multip]e-notch filter. Simulation results are presented to compare the performance of the proposed algorithm with the usual LMS algorithm and another variable step algorithm.

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